Klipsch sells replacement autoformers for a very modest price, something like US$25 or so. This is old information going back to the time when I was twiddling with the Klipsch Chorus loudspeakers - I also posted my results in more detail in the Klipsch forums, but that was something like four or five years ago. The performance of little Klipsch autoformers is excellent - the biggest downside is the taps are 3 dB apart, which is pretty wide spacing.

These days, I'd commission a special transformer from Bud Purvine or Dave Slagle, who both do terrific work that sounds first-rate. The transformer can be quite small, since they don't need to handle any current below 200 Hz, which in turn allows for a very small core size. As mentioned earlier, the turns ratio is the same the voltage ratio, and the impedance ratio is the square of the turns ratio.

This provides a load that is very close to resistive for the crossover network, and the compression driver sees a low source impedance both in the passband and in the band-reject region as well. Much better sounding than L-pad attenuation, since the impedance bumps are isolated from the crossover.

Originally posted by Keith Taylor Regarding magnet materials, Englishman John Watkinson has written on the subject in Electronics World magazine and in a more abreviated form at www.celticaudio.com/ Go to "technical articles" and read the last two paragraphs of the PDF called "putting the science back into loudspeakers" Mr Watkinson is no fool, being a fellow of the AES, the author of several books on digital audio and television and runs a company offering training in these and other subjects.

He seems to make a basic destinction between magnetic materials that are conductors and those that are insulators (ferrite) He suggests that in ferrites the magnetic domains move around when they are interacting with the coil flux and its attached load. This movement is not a linear process but rather is granular or noise like. The assumption seems to be that a ferrite magnet speaker is going to have noise sidebands associated with everything it reproduces. In the EW article he invited disbelievers to wrap some turns of wire around the ferrite magnet of a speaker being driven and look at/listen to the noise signal.

Keith

Well, there are parts of the article that are controversial, but I fully agree with his description of the sound of lossy codecs:

Quote:

Even at high bit rates, corresponding to the smallest amount of compression, it was obvious that there was a difference between the original and the compressed result. The dominant sound sources were reproduced fairly accurately, but what was most striking was that the ambience and reverb between was virtually absent, making the decoded sound much drier than the original.

What was even more striking was that the same effect was apparent to the same extent with both MPEG layer 2 and Dolby AC-2 coders even though their internal workings is quite different. In retrospect this is less surprising because both are probably based on the same psychoacoustic masking model. MPEG-3 fared even worse because the bit rate is lower. Transient material had a peculiar effect whereby the ambience would come and go according to the entropy of the dominant source. A percussive note would narrow the sound stage and appear dry but afterwards the reverb level would come back up. An opportunity arose to compare the same commercially available recording on CD and MiniDisc and the MD version was obviously inferior. All of these effects largely disappeared when the signals to the speakers were added to make mono which removes the ear's ability to discriminate spatially.

The effects are not subtle and do not require "golden ears". We have successfully demonstrated these effects to an audience of about 60 in a conference room on more than one occasion; hardly the ideal listening environment, but all heard it. One of us (Watkinson) was asked in one demonstration if this was only relevant to classical recordings so the demonstration was repeated with a Bruce Springsteen recording and again all heard the difference.

We are forced to conclude that because of the phenomena described here, audio codecs have reached the market, which produce audible artifacts even at high bit rates, despite exhaustive subjective testing. When one examines the results of any subjective compression test, it becomes clear that the type of loudspeakers used would have been those having the shortcomings mentioned above. As a result these subjective tests are invalid because the masking of the legacy speakers was masking the coder being tested.

We must conclude that whilst compression may be adequate to deliver post produced audio to a consumer with mediocre loudspeakers, these results underline that it has no place in a quality production environment. When assessing codecs, loudspeakers having poor diffraction design will conceal artifacts. When mixing for a compressed delivery system, it will be necessary to include the codec in the monitor feeds so that the results can be compensated. Where high quality stereo is required, either full bit rate PCM or lossless (packing) techniques must be used.

This is exactly what I hear when I listen to lossy-compressed digital on the Ariels - a source-modulated ambient impression that shuts on and off, and is in general rather dry and "electronic" sounding. As the lossy compression is removed, and the bit depth increased from 16, up to 20, then up to 24 bits, the spatial impression becomes progressively more natural and less "processed" sounding. I also find that the quality of the amplification - in particular, use of Class A and subsequent freedom from switching artifacts - has a strong effect on the audibility of ambient impression. Class AB amplification, whether in op-amps (which is almost all of them), or power amplifiers, can quite noticeably degrade the realism of the spatial impression, regardless of loudspeaker.

These spatial effects are clearly audible in 2, 4, and 5-channel playback systems, although in multichannel systems, there are additional perceptual artifacts (from lossy compression) that translate into a sense of fatigue and a hard-to-describe unnatural quality. My experience with multichannel is that it requires higher standards (in the transmission channel, amplifiers, and loudspeakers) than 2-channel for long-term fatigue-free listening. This would mirror the experience with the mono-to-stereo conversion - stereo, to the surprise of its advocates in the late Fifties, turned out to require higher playback standards, and is a less forgiving system than mono.

As shown in earlier references, the Ariel was specifically designed to have a rapid time-decay signature, although it is not a "linear-phase" loudspeaker. My goals with the new speaker are similar, although I am now looking for a 10~15 dB increase in headroom. The challenge with multichannel is sufficiently great I am not addressing it at this time.

The 12" Aquaplas JBLs had a lot going for them- some had very wide bandwidth and they all had monster motors to drive those heavy cones.

Just wonder if Alnico is the best place to spend the money on. What about cone design & material? SS uses slice paper cone. Seas uses some kind of coating - Nextel. What about Hemp cone on a good motor like lambda TD? Wouldn't this be a better cost vs benefit option?

Alnico is worth the money - at least in the context of high-quality sources and amplification. It's not one of those silly "better-sounding-cables" distinctions - it's more like going from 16 to 20 bits, or a much-improved dither algorithm. For those of you into the triode scene, I find it comparable to the difference between a good-sounding 300B and an EL34 - it isn't all that subtle. For the digital cats, it sounds like the difference between UV22 dithering and primitive, undithered recordings. Or reducing jitter at the DAC chip. Less grain, more music, more vivid tone colors.

Now, as for field coils vs Alnico, I dunno. They sound different (I think) but it wasn't anything that jumped out at me. I'm not sure one is better than the other.

The sonic differences don't sound like a change in cone material, or suspension. It doesn't really sound like a speaker coloration in the usual way you'd expect. It sounds a lot more like improving the electronics, which is why I suspect it has something to do with the IM distortion spectra of the speaker in the region where VC inductance modulation is starting to make a difference. But that's only a guess. The studio-monitor pro world, though, has strongly favored Alnico when given a choice, and they are very familiar with the sound of different ADCs and DACs.

With compression drivers, there's a lot more controversy - and that could be because the pole pieces are already close to saturation anyway, and the type of magnet behind them is less significant.

Just wonder if Alnico is the best place to spend the money on. What about cone design & material? SS uses slice paper cone. Seas uses some kind of coating - Nextel. What about Hemp cone on a good motor like lambda TD? Wouldn't this be a better cost vs benefit option?

Considering the upper frequency response I wonder how much better the cone material can be on the TD drivers..............

I have heard both good and bad about hemp and that actually the real "pot" hemp is pretty crap for a speaker cone.

Not too long ago i had a couple pairs of 123a JBL woofers running full range in an OB. The high frequencies were AWFUL, but the midrange and lower midrange were amazing. Acoustic guitar and tenor sax never sounded so good.

I am thinking about picking up a pair of the TD15m to listen to full range in an OB. Although it is a low excursion driver, (as was the JBL), I think it will work well. Does anyone have an opinion on the overall sound quality...and the improvements acheived in utilizing the Apollo motor over the standard?

Originally posted by nickmckinney If you wanted Alnico it can be done but I imagine the price tag would be outrageous for 99% out there as you need a massive amount of steel to complete the circuit. Plus a large enough chunk of Alnico isn't cheap either.

Well I shall remove my foot from my mouth, I just learned its expensive but not really that expensive now thanks to the Chinese. Lemme start looking at what we can do..............

- a fairly sized room – no need for ceiling heights intended for giants nor for the Vatican
- no special dampening of that room – give the famous "pile of cushions trick" a try once you'd like to take a nap – not so much needed for the measurement itself
- a decent microphone at a decent stand – if you are in name dropping buy a Microtech Gefell or a B&K - if not, basically every brand (and unbranded) > € 150.- / electret / condenser will do – they will offer sufficient impulse response and FR – besides it can be equalised easily - isn't that important here (if you have skills - a € 5.- capsule will do – though I wouldn't recommend to go that path)
- a decent soundcard with mic inputs (24bit is a good idea though not really necessary) or a external mixer that offers an appropriate mic input. Most of the above mics do need phantom power.
- PC in an other room or relatively quiet notebook – no turbo booster rocket thing needed
- A power amp and some decent cables - nothing high-end needed here at all
- ARTA (free-) Software – there are others as well – no need to spend a fortune as processing of data is basically all the same – invented many, many years ago – some more options here some more beautiful colors there
- something – like a stool - to put your speaker on if you already have done the work out for your biceps this day

Well, the most crucial to me is a good mic-pre and a good mic if you have some extra bucks to spend. Up from the Behringer to the NTE and Earthworks M30. Some types do not need any phantom powered mic-in at all.
What I use? Almost all my measurements were taken with the Earthworks QTC1 attached to a standard mic stand and the Mackie Onyx 400F.

The big trick on CSD compared to other forms of measurement is that you truncate measured data.
As long as ONLY data is processed that has not been spoiled by room reflections you will obtain a very clean measurement of the speaker itself.

Obviously no other (strong) noise sources should be present as well. This isn't sooo hard to do, as CSD graphs are usually only down to –25dB – meaning, that your ambient noise level may be up to around 65dB in case you measure at 90dB (more quiet is better of course to have some margin left). That should be no problem measuring in the desert of Sahara or in the desert of NY

To calculate what is realistic in terms of data not spoiled by room reflections we take a look at the graph below.

Each hard boundary will create reflections of the original sound source – the speaker we'd like to measure.
Not shown in the picture – but equally important – are also the reflections coming from the floor and from the ceiling as well as from the wall behind the listener – the microphone in our case.
Whatever boundary creates the reflection "seen" first by the mic will be the limiting factor for the time we have received unspoiled sound from the speaker itself.

Obviously – if the room is large and the speaker AND the mic are far from any reflecting surface, we get unspoiled sound for a longer "time window" .
At domestic rooms you are usually limited by the ceiling height. If – for example – you place the speaker and the mic at a height half the room height, the reflections from ceiling and bottom will arrive at exactly the same time .

Now lets do a simple calculation and train the muscles between our ears:
For a ceiling height of 2,4m a maximum of 2,4[m]/330[m/sec] = 0,0072727272...sec = 7msec of reflection delay can be achieved.
From the speaker 1,2m up to the ceiling (OR down to the bottom) and 1,2m back to the mic – in our example - makes a 2,4m in total.

For a possible time window of 7msec in this example, your room has to be at least 2,4m high 2,4m wide, 2,4m deep AND you will have to place speaker and mic precisely in the center.

Usually we don't want to place the mic flush to the speaker so the reflection free "time window" is slightly lower (train your brain if you recall trigonometry from school) – but a 3msec to 5msec "time window" is always possible.

What about reflections of the mic stand?
Ever heard an beautiful echo in the mountains? - well its quite different from echos created by a forest at some distance.
But echo of a single tree I never have heard.

To be fair, there IS sound reflection at ANY hard surface – but geometry of that surface and the absolute reflecting area translated into a room section (Raumwinkel in German) seen by the mic together with frequency dependant diffraction don't leave too much to worry about.

Once you always see the same patterns at measurement of different speakers then it may be worth to explore – but until now – I had luck - and why shouldn't you?